How to test series ONLY

USB Headset Measurement

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How do you measure USB headsets? In this short video, we demonstrate how to measure USB headsets using SoundCheck. You’ll see how to configure digital input and output channels, set up the headset in the hardware editor, and perform measurements including left and right earphone frequency response, THD, sensitivity, and microphone performance. This method can be used for any USB audio device, making it easy to accurately evaluate both playback and recording performance.

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Video Transcript: How to Measure USB Headsets

Today I’m going to demonstrate how we can measure a USB headset using SoundCheck. We’ll test both the left and right ears and the microphone.

SoundCheck supports both analog and digital inputs and outputs, so to test a USB headset, all we need to do is plug it into our computer’s USB port and configure a new digital channel in the hardware setup, then we can test it just like a conventional headset. 

Let’s review our setup. We’ll use a head and torso simulator to capture the headset’s response, and I’m powering that with a SoundConnect 2 power supply. The microphones within HATS transmit the signal to the computer via the AmpConnect 621 Audio Interface, and that also has a built-in amplifier to power the HATS mouth, which we need for the microphone test.

First, I’ll configure the USB headset in SoundCheck. We do this in the Hardware Setup. You can see here we already have the input and output channels of the AmpConnect 621 configured  – it’s full plug and play so that happens automatically. 

I’ll plug the USB headset into the USB port of my laptop. In the hardware editor, I’ll right click to create an input channel. I’ll set it to use the Windows WDM driver, and now it shows me all the available devices. We see the headset mic there – let’s set the type to digital. Since it’s digital we don’t need to enter anything in the calibration field. I’ll leave the sampling rate as the default, as my audio interface is using this value. We’ll also do the same for the outputs. We’ll set the driver and pick the device, there’s 2 channels so we’ll set these to left and right, and again it’s a digital device. 

First we’ll measure the headphone response, and we’ll use the USB headset test sequence that’s available for free on our website. 

When we open our sequence, it prompts us to re-link the signal paths used in the sequence. Let’s add them to system calibration.

And now we need to assign hardware channels and calibration files. It’s warning me that the signal paths I just added don’t have associated hardware channels and calibration data, so we’ll click through all of those so we can add them. Here’s our signal paths.

We’ll assign signal paths digital out 1 and 2 to USB Headset out L and R. Since these are digital outputs the calibration is unity digital out

Now we’ll do the same for the input headset sequence. We’ll set Digital In 1 to the USB headset and calibration to unity digital in

Let’s just do a quick check and make sure all our signal paths are correct. It’s looking good

So now we’re all set up and ready to measure. 

Let’s hit start. 

The sequence plays our test stimulus, and it gives us the left and right frequency response, and also the THD for the left and right earphone. Over here we see the waveforms, the left and right recorded response, and down here we calculate our average sensitivity. Sensitivity is a useful production line test to confirm that the average level across a bunch of headphones is reasonably stable. If the sensitivity swings dramatically in either direction from sample-to-sample, it can indicate a quality problem in manufacturing. We also show the left/right tracking – that’s the difference in sensitivity between the left and right ear over the frequency range.

Now we have some results, let’s take a look behind the scenes at what our test sequence is doing.

We’re using a Stweep that sweeps from 20 kHz down to 100 hertz at 12th octave resolution. This is a compound stimulus with a very short, 1kHz pilot tone, before the Stweep. The pilot tone isn’t analyzed, but we use it in this frequency shift step here  to align the time and phase of the response signal with the stimulus before we analyze it.

This feature ensures accurate analysis. It compensates for any unknown phase differences that are introduced from the stimulus and response being played through different hardware devices – the stimulus is played through the laptop via USB, and the response is recorded through our audio interface.

Next, our analysis step compares the stimulus waveform with the frequency-shifted version of the response waveform to get the fundamental. There’s also a post-processing curve averaging step that calculates the average sensitivity.

Lastly, the display step that defines what graphs and results are shown. So that’s the earphones, now let’s test the microphone. We’re already connected and set up, so I’ll tell the sequence to continue and measure the microphone.

So that’s the earphones, now let’s test the microphone. We’re already connected and set up, so I’ll swap over to the microphone test sequence and measure the microphone.

We heard our test stimulus coming out of the HATS mouth, and being recorded by the headset microphone. And here we can see the frequency response and sensitivity of the microphone.

And if we look at the sequence, we can see it’s very similar to the headphone. We’re using an 8 Khz to 100Hz test stimulus, again with a trigger tone to align our stimulus and response waveforms. We have our frequency shift calculation, just like before. Then our analysis step to calculate  frequency response, and also a post-processing step to calculate sensitivity. And of course we have a display step to determine what is shown on screen.As you can see, because SoundCheck supports any combination of analog and digital audio devices, it’s very simple to measure a USB headset. All you need to do is make sure you have it set up as a hardware device in SoundCheck, and use a compound stimulus and frequency shift step to align your waveforms before analysis. You can measure any other USB connected audio device in the same way. Check out our website to learn more and download the SoundCheck test sequence.

Hearing Protection Measurement

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How do you measure hearing protection devices? In this short video, we demonstrate how to measure hearing protection using SoundCheck in accordance with the ANSI S3.19-1974 standard. You’ll see how to set up a head and torso simulator, calibrate the system, and perform repeated unoccluded and occluded measurements using pink noise to accurately calculate attenuation and Noise Reduction Rating (NRR). This method can be used for a wide range of hearing protection devices to ensure reliable and standardized performance testing.

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Video Transcript: How to Measure Hearing Protection

In this short video, we’ll measure the performance of hearing protection devices and calculate the Noise Reduction Rating using the ANSI S3.19-standard.

Our setup is simple. We have a head and torso simulator to wear the hearing protectors and capture the signal, a speaker to play the test stimulus, and a calibrated reference microphone for speaker calibration. The stimulus and response signals are transmitted to and from the computer using the AmpConnect 621 audio interface which also contains an amplifier to power the speaker. I also have a SoundConnect 2 to provide 200V polarization voltage to the ears since I’m using an older HATS. The newer versions have pre-polarized ear simulators so they can be powered directly from Listen’s audio interfaces.

We’ll play a pink noise test signal and capture the response with the ears unoccluded. We’ll then put on the hearing protectors and make the occluded measurement. We’ll re-position and re-measure several times and take an average to account for fit variations. We’ll then compare the occluded and unoccluded data to calculate the noise reduction ratio, or NRR.

My reference microphone is already calibrated, and connected to input 1. Now  I need to calibrate the ear simulator input signal paths and source speaker in the calibration editor.

Let’s go to  Setup > Calibration and go to Input Signal paths > Ear Sim L. I have connected this to Input 2, now I’m ready to calibrate the ear. I’m going to use a B&K 4231 calibrator with the coupler UA 1546, hit the calibrate button and save the sensitivity value. Now we’ll repeat that for the other ear, Ear Sim R which is connected to Input 3.. 

Next, I need to calibrate the source speaker. I’ll place the reference mic as close to one side of the hearing protector test fixture as possible. We’ll navigate to the output signal paths tab inside the Calibration editor, and select Source speaker. I’ll make sure it is assigned to Output 3 because that’s the amplifier output of the AmpConnect 621 where we have it connected. Now I’ll click Calibrate using Reference mic as the input. We’ll use a level of 80dBSPL from 10kHz-100Hz, the same frequency range that we’ll use in the sequence. It’s not quite flat, so we’ll iterate a couple more times until the EQ’d response is flat. This looks good. Now, everything is calibrated, we’re ready to run the sequence. 

Let’s hit start.

First, the sequence wants to know how many trials I want to do. We’re going to make this measurement several times, removing the DUT and placing it back so that we can average the response to account for fit variation. We’ll run it 5 times for the purposes of this demo.

First, we’ll capture the unoccluded response – this means measuring the sound levels without any hearing protection in place. The test sequence will play pink noise through the speaker and record the sound spectrum at both ears of the fixture. Let’s hit enter.

Let’s save this measurement.

Now the sequence is asking me to place the hearing protection device onto the test fixture and we’ll continue to repeat the test to measure the occluded response. Now we’ll just follow the instructions and do it four more times.

Next, the sequence performs postprocessing steps on these unoccluded and occluded measurements to calculate the attenuation at different frequencies, average attenuation, standard deviation and finally the Noise Reduction Rating (NRR) – you can see its 26.6 on the left and 32.1 on the right. These measurements provide a complete picture of the device’s performance. 

And that’s it! Using this pre-written sequence – which you can download from our website – it’s quick and easy to  accurately measure attenuation and determine the Noise Reduction Rating of any hearing protection device to the ANSI standard.

Hearing Aid Measurement

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How do you measure hearing aids? In this short video, we demonstrate how to test hearing aids using SoundCheck with standard ANSI S3.22 measurement techniques. You’ll see how to set up an anechoic test box and coupler, and configure SoundCheck for key measurements including frequency response, input/output linearity, OSPL 90, and release time. This method can be used to accurately evaluate hearing aid performance and ensure compliance with industry standards.

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Video Transcript: How to Measure Hearing Aids

Today,  I’ll demonstrate a hearing aid measurement using SoundCheck with some of our standard hearing aid test sequences. We’ll use some of the tests outlined in the ANSI S.22 hearing aid test standard – frequency response, input/output linearity, OSPL 90, and release time. 

First let’s take a look at our hardware setup.

It’s a little more involved than something like a loudspeaker measurement because hearing aids have some unique test requirements. For a start, we need an anechoic test box  – this one’s a B&K 4232 but you could use anything similar. It has a speaker integrated inside, which we’ll play our test signal out of. We’ll put our hearing aid in this coupler with a ½” pressure microphone. We also have a SCM reference free-field microphone which is also going to go in the test box. The ½” mic needs LEMO power, so it’s connected to a SoundConnect 2 power supply, and the mic input then goes into the AmpConnect 621 Audio Interface. The AmpConnect also provides power for the SCM mic and amplification for the speaker in the test box. 

If I were running the complete suite of ANSI S.22 tests, I would also need a DC power supply and current monitor but since the tests we’re running today don’t require voltage regulation or current monitoring, we’ll just power the hearing aid from its internal battery.  

I’ve already calibrated the source speaker located inside the test box using my SCM reference microphone pointed at the source speaker.  The coupler is configured with a ½” microphone capsule, and we’ll attach our hearing aid to the coupler’s input – we have some tack in here to get a good seal. Now we’ll put it in the test box so its microphone port is located at the same position I had the reference microphone.  Let’s close the box, and we’re ready to test.

Let’s start with Frequency Response. The hearing aid should be set to its reference test gain. We’ll play an equalized stepped sine sweep from 10 kHz – 100 Hz at a level of 60 dBSPL through the test box speaker, and analyze the output of the hearing aid with the Heterodyne algorithm to produce a frequency response. Next, the HFA,  or High Frequency Average is calculated by averaging the response values at three frequencies – 1000, 1600, and 2500 Hz. We then subtract 20dB from the HFA, and two post processing steps are used to find the upper and lower frequency points at which the response curve intersects the HFA -20 value. These are the high and low frequency cutoff points. Limits can be applied to all of the curves and values of this sequence. Let’s run it.

And here you see the Fundamental curve falls outside of the upper and lower limits at a few frequency points, the low frequency cutoff is 217 Hz and the high frequency cutoff is a little over 5 kHz.

Our next test is input/output linearity. We’ll leave the hearing aid at reference test gain, and use an amplitude sweep from 50-90 dBSPL played through the test box speaker. This is repeated at four frequencies (1 kHz – 4 kHz) in 1 kHz increments, and the output of the hearing aid is plotted vs. the input on the graph. This test is useful for characterizing any automatic gain control or compressor in the hearing aid,  and is used to attenuate the hearing aid’s output above a certain level so as to not further damage the user’s hearing. Let’s run it.

The effect of the automatic gain control is clearly visible at all four frequencies.  The x-axis represents the stimulus level and the y-axis the output level from the hearing aid.  We can see that the gain limiting kicks in when the input is in the 60-75 dB range, depending on the frequency and note how the overall level of the 4 kHz curve is 10 dB lower than the others because the frequency response of the hearing aid is around 10 dB lower at that frequency relative to the others.

The OSPL 90 test measures the hearing aid’s performance with an input signal of 90 dBSPL.  For this test we need to set the hearing aid gain to its maximum. An equalized stepped sine sweep from 8 kHz – 200 Hz is played at a level of 90 dBSPL, and a broadband rms response curve which includes harmonics as specified in the standard, is analyzed from the recorded time waveform of the hearing aid’s output. Next, the HFA (High Frequency Average) is calculated by averaging the values at three frequencies (1000, 1600, 2500 Hz), and the Max OSPL is calculated by finding the maximum point on the broadband response curve. Let’s take a look.

The OSPL-90 curve looks much like the frequency response curve from our earlier measurement and considering that the input to the hearing aid is a calibrated sweep at 90 dB, the amount of gain in our measurement is between 25 and 40 dB across most of the sweep range. The high frequency average from this measurement is 126.7 dB and the maximum value on the curve is 129.7 dB.

Lastly we’re going to measure the release time of the hearing aid. Again, we want the gain set to max, and we’ll play a  2 kHz sine tone at 90 dBSPL for 1 second, then immediately drop it to 55 dBSPL for 2 more seconds. A band limited time envelope to improve the measurement signal to noise ratio – from 1.5-2.5 kHz –  is calculated and then run through a post processing step, to calculate the release time – that’s the time the coupler SPL remains within 4 dB of the steady value for the 55-dB input. [run test]

Here we can see that the hearing aid output initially drops down to around 85 dB but after 45 ms the release time has stabilized and the hearing aid maintains a steady output level of 115 dB.

So that’s four quick examples of tests from the ANSI S.22 hearing aid standard. Of course, once you have the equipment for these, there are many other hearing aid tests that you can run using the same test setup, and you can configure tests for the complete ANSI S.22 standard in SoundCheck. In fact we’ve already done it, and we sell the test sequence package with all 20 measurements as an add-on to SoundCheck so you don’t even have to program it yourself! Please contact us to learn more.

ANC Headphones Audio Measurement

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How do you measure ANC headphones? In this video, we show how to measure active noise cancellation (ANC) performance using SoundCheck. You’ll learn how to test unoccluded, passive, and active responses with a head and torso simulator (HATS), an AmpConnect 621 audio interface, a SoundConnect 2 power supply, and pink noise source to calculate total noise attenuation. This simple setup can also be expanded for real-time ANC analysis or used to measure THD and distortion caused by ANC circuits.

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Video Transcript: How to Measure ANC Headphones

Today I’m going to show you how to measure active noise cancellation performance, or ANC, of headphones.

The set-up is simple. I’m using SoundCheck to generate the test signals and analyze the results. I have a head and torso simulator to capture the ear’s response. I’m using an AmpConnect 621 audio interface which also provides amplification for the speaker… and I have a SoundConnect 2 microphone power supply to power the mics in the head and torso simulator. If I was using a newer HATS with IEPE mics, I wouldn’t need this – I could power it directly from the AmpConnect. I’ve got a basic speaker here to generate the pink noise that we’ll use to measure the noise cancellation, and I have a nice pair of noise-canceling headphones.

We’ll play pink noise from this speaker. First we’ll measure the unoccluded ear response, then the passive response. That’s with the headphones on the ear but not turned on. Then we’ll turn on the ANC and measure the total attenuation – the full effect of the active and passive isolation.

Let’s run our test sequence. First we’ll take the unoccluded measurement without headphones. You’ll hear the pink noise playing, and we’ll see our measurement on the graph.

There we go…

Now the sequence pauses while we put the headphones on.

Next we’ll play the pink noise. I’m going to adjust the headphones to establish a good seal. Typically you would make this measurement 5 times and average it to account for variation in fit, but I’m just going to do it once in this demo.

Now let’s continue.  [pink noise plays]

So now we have the occluded measurement – that’s telling us how much sound the headphones are passively attenuating just from their materials and geometry.

Now we’re going to turn on the ANC and let the sequence continue….

Here we have the measurement with ANC turned on.

And you can see on the top graph here, we show the sound pressure level for the unoccluded measurement, the occluded measurement, and the measurement with ANC turned on. And this bottom graph here uses a simple calculation to show the passive, active, and total noise attenuation of the headphone.

That’s the test in its simplest form. You can get more fancy with your speaker setup. For example, instead of using just one source speaker, you could use multiple equalized source speakers to create a more realistic spatial background noise environment. You could also measure both ears at the same time. If you do this, you can use SoundChecks multichannel RTA to check for fit – you just measure the two channels simultaneously and make sure the left and right are showing similar responses. This saves time as you don’t need to make multiple measurements.

You could also use the multichannel RTA to look at how noise attenuation changes in real time. You’d need an external reference measurement microphone positioned here, outside the headphone, and you could play complex signals, like real recorded background noise from an airplane and watch in real time how the noise attenuation changes. This gives a more realistic representation of how the ANC circuit responds to dynamic signals.  

This set up is good for much more than just measuring active noise cancellations. We’ve done a lot of research on audio distortion caused by active noise cancellation. With the same setup, but different test sequences, you can measure THD, intermodulation distortion, time-variant distortion and more. Check out the technical papers on our website to learn more.

Smart Glasses Audio Measurement

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How do you measure smart glasses audio? In this video, we demonstrate how to test smart glasses using SoundCheck with a head and torso simulator and Bluetooth interface. You’ll see how to measure speaker frequency response, total harmonic distortion (THD), and left/right balance, as well as how to evaluate the built-in microphone using hands-free profile (HFP) measurements. This simple method can be applied to smart glasses, AR/VR headsets, or any Bluetooth device with speakers and microphones.

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Video Transcript: How to Measure Smart Glasses

It’s easy to make audio measurements on smart glasses using SoundCheck. Let me show you how.  As you probably know, most smart glasses and VR/AR headsets contain small speakers, usually embedded in the arms close to the ears. They also have microphones also hidden in the frames. This means we can make audio measurements on smart glasses pretty much the same way we do with a Bluetooth headset. 

Let’s start with our setup. You need a head and torso simulator, or at least a test head. This lets you position the glasses as they would be worn in real life, and record the sound at the ears. If you want to measure the microphone as well as the speakers, you’ll need one that includes a mouth. Alongside this, we’ll use an AmpConnect 621 audio interface. This transmits the signal from the microphones into the laptop, which is running SoundCheck, and it also provides amplifier output to power the HATS mouth. We’re also going to need a SoundConnect 2 microphone power supply because I’m using an older HATS that needs 200 Volts polarization power. If you have a newer version you can power it directly from the 621. And I’m going to use a BQC bluetooth interface to communicate with the smart glasses.

We’ll use two basic test sequences that you can download from our website. Let’s run the sequence and take a look at the results, then I’ll explain what we’re doing.

First the sequence is prompting me to pair my device. I’ll put it in pairing mode and say ‘yes’.

We can see it in the list here…

Now it’s telling me to put the smart glasses on the head and torso simulator… lets do that

And we’re ready to make our measurement so I’ll hit enter to continue

That was the test signal, a Stweep, which is a special type of stepped sine wave.

Now we can see our results. On the top graphs we have the left and right frequency response – you can see they are pretty well matched. The response is rolling off quite a bit in the low end, so anything below about 500Hz in all of these measurements will be very noisy.

Here we have the total harmonic distortion of both the left and right speakers..

And down here at the bottom we have the left/right balance which you can see is pretty flat up to about 8kHz. Above this, we see some discrepancy as the responses of both ears have more peaks and dips.. This is very sensitive to positioning, so it’s important to make sure the glasses are correctly positioned on the head. It’s also good to make multiple measurements, completely removing and replacing the glasses to see how the balance changes.

Now let’s take a look at the sequence and I’ll show you what we did.

We start with our stimulus step, which is configured to sweep from 20kHz to 100Hz. Then we have some message steps – that’s where it was prompting us if we wanted to pair the glasses using Bluetooth and confirm once it was paired and placed on HATS. Then you can see we play the stimulus and record the response.

Now we have some post-processing going on. Because the Bluetooth output and AmpConnect input don’t use the same master clock for sampling,  we have to resample and frequency shift so that the stimulus and response are aligned for analysis. I’m not sure if you heard the short tone at the beginning of the test signal –  that was a ‘trigger tone’ that lets SoundCheck precisely align the waveforms by playing a signal within the pass band of the device. Once we’ve done this, we can run our Harmonictrak analysis to calculate frequency response and distortion, and also another post-processing step to calculate the left/right balance. Finally, we have a display step that shows all these results on screen.

These smart glasses also feature a microphone for making calls and recording, so let’s now measure that. We’ll use exactly the same setup with a different sequence. This time we’re going to play the stimulus from the mouth, and capture the recording using the microphone.

Let’s run the test… 

We’re pairing it again and we’re switching the Bluetooth profile to HFP – hands-free profile as this is what you’d typically use for making calls. We made the speaker measurement using the A2DP profile as that’s what you would typically use listening to music.

Now it’s going to play the test signal… and output the frequency response… and we can see it’s pretty uniform  from about 300 Hz to about 7k or so,   then it rolls off.

Let’s look at the sequence to see what we did here. 

Here we have our stimulus, sweeping from 8kHz down to 100Hz, which aligns with the HFP Bluetooth profile that we’re using for this measurement.

We have our message steps, the same as before, to connect via Bluetooth 

And here we have our stimulus and acquisition step, you can see we’re playing out the mouth simulator.

And after we’ve done our frequency shift and resampling, we calculate the frequency response of the microphone  and display that on the graph.

So, as you can see, testing smart glasses using SoundCheck is quick and easy – just a couple of minutes to run the tests. And these are pretty basic tests – if you want more extensive measurements you can simply modify the sequence to add any other results that you want. With a little extra equipment, you could also make measurements with background noise, or make audio leakage measurements. You can download these test sequences for free on our website, and they’ll work with any bluetooth device with a speaker and microphone so not just smart glasses but headsets, VR and AR goggles etc.. Check it out, and call us if you have any questions.

Car Audio Measurement – Part Two

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In this second part of our car audio measurement demo, we’ll be covering how to measure maximum SPL at the onset of audible distortion and detect impulsive distortion using advanced SoundCheck sequences. You’ll see how to identify buzz, squeak, and rattle with perceptual Rub & Buzz, transient distortion, and crest factor analysis.

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Here are some additional resources:

Video Transcript: How to Measure Car Audio – Part Two

Today, I’m going to continue on from our last automotive video where we measured frequency response, spectral uniformity, THD and IM distortion. This time we’ll  demonstrate Max SPL at the onset of audible distortion  and impulsive distortion. Let’s make some measurements.

Again, I’m using the test configuration recommended in the AES TC-AA white paper on automotive measurements. We’re using a precisely positioned 6-mic array in the driver’s seat to represent our listener. This is connected to  an AmpConnect 621 Audio Interface. Our laptop is also connected to a measurement-grade Bluetooth interface  which we’re using to send the stimulus signal into our car’s head unit. If you want to know more about the test configuration, you can download the AES white paper, and we also have full details on our website. The inputs and outputs of the test system are already calibrated, as is the  audio interface, so we’re ready to measure.

Maximum SPL is one of the measurements outlined in the TC-AA proposal, and it’s important because it lets you compare maximum sound levels of infotainment systems. It’s defined as the maximum sound pressure level that the infotainment system can reproduce, and it combines all the components of the system – the speakers, amplifiers, power supply and so on.

The simplest way of evaluating this is to measure the overall Max SPL and Max SPL spectrum regardless of distortion. In these tests we set the car’s volume control to maximum and measure overall Max SPL and SPL Spectrum simultaneously using a 30 second broadband 20 to 20 KHz monophonic pink noise stimulus at minus 12db FS . This is quite a basic measurement and I’ve demonstrated it in other videos, so we’ll just briefly discuss the results here, then I’ll demo a more advanced measurement. 

Here we see the individual microphone measurements and the average, the black line. The overall level of 103dBC, regardless of distortion, isn’t too loud but maybe it might be limited by design to prevent severe distortion at low frequencies from clipping the amplifier. 

What I’ll show you now is a Maximum SPL measurement at the onset of audible distortion. More volume is only good as long as the output sounds clean, so it makes sense to define the Maximum SPL based on when the sound quality starts to degrade.

This is a more complex test sequence and takes longer to run because you need to repeatedly play the stimulus until you reach distortion across a range of frequencies – similar to how we measure Max SPL in other applications. The good news is that this process can be highly automated in SoundCheck.

Our test sequence plays the stimulus multiple times for each test frequency, increasing the level in 3dB steps until the THD exceeds a predefined level in percent. It then reduces the level by 3dB and increases it in smaller increments of 0.5 dB to more precisely find the THD threshold. The Max SPL is recorded at that frequency and the test sequence continues to the next frequency. The entire process is completely automated via the test sequence, making this measurement fast with minimal intervention.

This sequence actually takes about 20 minutes to run in total, so I’ve edited this recording to skip the repetitive bits. We’ll set our range from 50 Hz up to 500 Hz, and we’ll set the percentage distortion. I’m going to set that at 3%. That means that once the distortion level goes above 3% it will record it and move onto the next frequency.

We’ll start at minus 12dB. The first thing it does is an auto range and it measures the time delay in the system… Now it’s playing 50 Hertz…It’s auto ranging. In a moment, it’s going to show the first result.

All right, here we go.

So, we can see the 50 Hertz sine wave of the 6 microphones capturing it, and we can see that we’re around 3.4% distortion.

So now it’s going to lower by 3 dB.

And it’s going to start going up again.

So now we’re 114 dB at 2.4%.

Now it’s going in half DB steps to very precisely find the sound pressure level at exactly 3%.

I’m just going to skip quickly through the next few measurements until we start graphing results

OK. We got 50, 63, and 80 Hz. You can see on the bottom graph is the input level and on the top graph we have the maximum sound pressure level versus frequency, the blue curve, and the orange curve is the THD level for that sound pressure level.

Again, I’ll fast-forward through the rest until we get our end result – it’s just more of the same at a bunch of different frequencies

OK, here’s our final result. The top graph shows the Maximum SPL and corresponding THD, and the lower graph shows the maximum stimulus level plotted when the THD reaches 3% with the head units’ volume control set to max.

At some frequencies, especially higher frequencies, for example 400 Hz and 500 Hz, the maximum input level is reached before the system distortion reaches 3% THD. The stimulus level is typically limited by the lower frequencies where there is more distortion.

So that’s it for system performance measurements. Now let’s look at system integrity.

Impulsive Distortion is a useful system integrity metric. It detects any audible buzz, squeak and rattle that may be caused by the speakers vibrating the door panels, loose wire harnesses, etc. These noises degrade the listener experience so it’s important to assess the vehicle thoroughly. It’s actually a good idea to make these measurements before you make the system characterization measurements, as this ensures that the infotainment system is properly installed in the vehicle under test

You can also use this test as an end-of-line check to verify vehicle integrity.

There are several methods of measuring impulsive distortion. The AES White Paper suggests a crest factor measurement with a logarithmic sine sweep and a tracking high pass filter, to evaluate impulsive distortion at 80 and 90 dBA. Sine waves have a crest factor of 3dB, and transients, such as rattling wires, loose fasteners, and noise, for example background noise, typically have a crest factor greater than 8dB. Generally speaking, crest factors over 3 dB, may indicate buzz, squeak or rattle. However, you have to be a bit careful as background noise, especially in a noisy factory, can result in similar Crest Factor results. You have to be very careful to minimize background noise and make sure that the results correlate with what you can hear.

I prefer to use perceptual Rub & Buzz and Transient distortion versus time instead, as they are less sensitive to background noise, and only highlight distortion sounds that are audible to humans. Perceptual Rub & Buzz uses masking curves to simulate the performance of the human ear to measure only audible distortion.  Transient distortion versus time measurement tracks individual events in the time domain to reveal any loose particles or rattling wires. It offers greater precision than frequency domain measurements since it tracks individual transient events.

I’ll measure both these parameters simultaneously, using a stepped sine sweep stimulus.

There’s a couple of frequencies there where I could hear some buzzing or rattling sound, I’m not sure if you could hear it. Let’s see if we pick that up in the measurement.

OK,  so here’s our recorded time waveform of all 6 mics.

And here… we can see the perceptual rub and buzz and indeed there’s a little bit of a peak going on at around 224 Hz and probably at a higher frequency here to around 340 Hz. These are the little buzzing sounds that we heard during the sweep.

Now, if we go to the bottom waveform graph, we can actually see if there’s any transients. Those are typically caused by things rattling around.

We don’t have much here – at low frequencies we don’t have a lot of output, so I suspect this is just background noise. So this car isn’t bad – just a little bit of Rub & Buzz and no transient artifacts.

I actually tried cranking the SPL up to 90dB to try to get some distortion at higher volumes, and here’s what I got…

Let’s just take a listen to the recorded response…. At this volume you can hear the distortion a lot more clearly.

Here we have the results of crest factor, Perceptual Rub & buzz and transient distortion.

The distortion is so obvious at this volume that it even shows up on the Crest Factor response – you can see the Crest Factor is above 8dB between 200 – 300 Hz.

It’s also much more obvious on the PRB curve – you can see a peak in the same point showing it correlates well. This is a little more sensitive than crest factor though –  we can also see a smaller, sharper bump at 600 Hz which is not perceptible using crest factor measurement, even though I could hear it during the sweep. This measurement technique is also considerably more reliable in a noisy environment.

This 3rd graph shows the time transients of the impulsive distortion measurement. You can see the constant background noise as a steady noise floor and impulsive distortion is highlighted as well-defined transient spikes. You can see transient spikes from 2 of the microphones around 11-12 seconds into the sweep. These are closer to the driver side door where a door panel is rattling at low frequencies, around 200-300 Hz, in the sine sweep. A neat trick with this measurement method is that you can actually listen to the recording without the fundamental, let me just play that back now… and that really highlights the audible distortion.

So that pretty much concludes our suite of measurements. There are of course many variations of these measurements as well as other things you could measure with the same physical hardware setup, but that’s all we have time for today. There are some more in-depth automotive audio testing videos on our website and YouTube channel, so if you want to learn more, check them out.

Car Audio Measurement – Part One

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How do you measure car audio? In this video, we demonstrate a complete car audio measurement setup using SoundCheck, based on the AES TC-AA recommended configuration. You’ll see how to test automotive infotainment systems inside a real vehicle using a calibrated 6-microphone array in the driver’s seat to capture spatially averaged data. We walk through measurements for frequency response (using both correlated and uncorrelated pink noise), seat-to-seat spectral uniformity, total harmonic distortion (THD), and intermodulation distortion.

Watch this Video Demo of Car Audio Measurement Techniques

Learn More about How to Perform Car Audio Measurements

Here are some additional resources:

Video Transcript: How to Measure Car Audio – Part One

Today I’m going to demonstrate in-car audio measurements, specifically frequency response, spectral uniformity and THD, using the test configuration recommended in the AES TC-AA white paper on automotive measurements.  

We’re using a precisely positioned 6-mic array in the driver’s seat to represent our listener. This is connected to  an AmpConnect 621 Audio Interface. Our laptop is also connected to a measurement-grade Bluetooth interface  which we’re using to send the stimulus signal into our car’s head unit. If you want to know more about the test configuration, you can download the AES white paper, and we also have full details on our website.

I’ve already calibrated the inputs and outputs of the test system by measuring the sensitivity of each individual microphone in the array and storing these readings in the software calibration menu. I also calibrated the audio interface to ensure proper response frequencies and levels, both in the analog and digital domains, so we’re ready to measure.

First we’ll measure frequency response using 80 dBA monophonic, correlated, pink noise on both channels. I’ve already calibrated the level, and we’ll play a short burst. Ideally the stimulus should be longer but I’ll still get a pretty accurate result. Let’s start the sequence. Enter the information it’s asking. OK, the overall level is 79.5 dB, quite close to 80, with a delay of 1.45 seconds which SoundCheck has measured and compensated for. What I’ve actually done here is a 1/3 octave analysis of each individual microphone, from 1 through 6, and you can see the difference due to location, reflections, and standing waves in the car. The black curve is the overall power average or spatial average of all six mics and represents what a human would hear.

Now let’s  measure the frequency response using uncorrelated pink noise. This randomizes the noise between the two channels rather than playing identical signals out of both. We expect to see some low frequency cancellation due to phase differences.

So let’s run the sequence… and we’ve got our results.

We can compare the results with uncorrelated noise to the results when I used correlated pink noise. Let’s bring them both onto the graph… The orange line is the correlated spatial average and the black line is the uncorrelated spatial average. We can see the uncorrelated pink noise has  about 3db less output than the correlated pink noise below 100 Hz and very little above 1KHz.

We can also look at the seat to seat spectral uniformity. Ideally each passenger in the car experiences the same high quality sound, but it’s hard to independently equalize the loudspeaker playback system for each seat in the car. Typically the driver’s seat is favored, the frequency response in the other seats is not quite as smooth.

I’m not going to do this now as it takes a while, but I’ll show you some measurements I made earlier. I repeated the first frequency response measurement with correlated pink noise in each of the car’s four seats and here you can see that the base performance in the rear passenger seats, the orange and green lines are not as extended or smooth as the front passenger seats.

To calculate the seat to seat spectral uniformity, We compare the individual seat response to the maximum and minimum responses to find the seat with the biggest difference. In this example both the left and right rear passenger seats have over 10 dB difference from the front seats at some

Frequencies. The lower the number the more uniform the frequency response is from seat to seat.

Now let’s make some distortion measurements. We’ll start with THD.

THD indicates the overall nonlinear performance of the device, and if it’s outside the limits it might indicate a poorly centered voice coil or mechanical defects. We measure total harmonic Distortion at 80 dBA,  this time using a stepped sign sweep from high to low with 12th octave steps.

Let’s run it…  As before, we power sum the spatial average of all six microphones to produce the THD power average curve, this black line here, which smooths out the standing waves and reflections picked up by the individual microphones. That’s not too bad –  if we look at the power average we’re getting roughly 2.2% from 50 to 10 kHz.

Next we’ll look at intermodulation Distortion. This is a good test for multi-way speaker systems with a crossover- if a small driver is trying to play back low and high frequencies simultaneously it’ll struggle to play both frequencies and create IM distortion. Here we use a two-tone stimulus with a fixed tone set to 50Hz, and a sweep tone that sweeps high to low from 20 kHz down to 150 Hz. I’ve already run a couple of sweeps to set the auto delay and to Auto Range the input gain for the best signal to noise ratio – this is important to ensure that the harmonics are above the noise floor of the measurement. Now let’s start the measurement. 

OK… The graph shows the fundamental and total Intermodulation Distortion of the second and third order intermodulation products, power averaged from the six microphone array. As you might expect, there’s more IM distortion at the lower frequencies, but you can see an increase around 700 Hz – that’s probably a crossover frequency in this car. If we take the spatial average of all six mics we get an overall IM distortion level of 1.4%.

That’s all we have time for here – check out part 2 for Max SPL and more distortion measurements!

Microphone Measurement

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How do you measure a microphone? In this short video, we demonstrate how to accurately measure a microphone’s frequency response using SoundCheck. You’ll see how to ensure an accurate microphone measurement by using two different techniques that separate the microphone’s response from the source speaker’s influence. We cover two key methods: the Equalized Speaker Method, which calibrates the speaker using a reference microphone for a flat response, and the Substitution Method, which compares the test microphone to a reference microphone to remove speaker effects.

Watch this Video Demo of Microphone Measurement Techniques

Learn More about How to Perform Microphone Measurements

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Video Transcript: How to Measure Microphones

The challenge with measuring the frequency response of microphones is that usually, the microphone’s frequency response is better than the source speaker used to play the test signal to measure it. This means you need to use measurement techniques that ensure you’re measuring the microphone and not the speaker. SoundCheck offers two great ways to do this.

The first is what we call the Equalized Speaker Method. First we measure the source speaker with a calibrated reference omnidirectional microphone that has a flat frequency response. We apply the reciprocal as a equalization  curve for the speaker at a constant sound pressure level, then measure the microphone under test or DUT in the same position as the reference mic.

The second method is the substitution method. Here, we first measure the source speaker with a reference microphone, then the source speaker with the microphone under test in the same location. Then we subtract the microphone under test result from the reference microphone to show just the characteristics of our DUT. Unlike the Equalized Speaker Method, the substitution method is not performed at a constant SPL which can result in a poor signal to noise ratio where the source speaker response rolls off at low and high frequencies significantly.

Let’s take a look at these measurements

Pre-written sequences for both of these methods come with the SoundCheck installation, so we’re going to use those today.

I’ll start with the equalized speaker method. I’m using an SCM measurement microphone as my reference microphone, and I’m using the AmpConnect 621 audio interface because as well as microphone inputs, it has an integrated amplifier to power my source speaker – a single driver, wide-bandwidth, Cambridge Audio Minx. This makes it easier to repeatedly position the microphones by pointing them at the center of the speaker.

First we need to equalize the speaker. We’re going to use SoundCheck’s built-in calibration sequence to do this.

It plays our stepped sine signal, measures the frequency response… we have a few reflections here. We take the reciprocal of the response and equalize the  output of the signal generator and measure again. You can hear its boosting the low frequencies instead of it being + or – 20 db its + or -1 db or less…. Because the speaker’s non-linearity it will be slightly different at different levels so we can equalize the equalization to further refine the EQ curve.

Here’s our speaker and we can see the EQ curve and the residual correction after EQ… and you can also see the phase here.

Now we’re all calibrated, let’s make a measurement. We use the same stepped sine sweep played  through the speaker, capture the frequency response, calculate the sensitivity at 1kHz, and display our results on a graph.

We also have a couple of arbitrary limit steps here – you’d want to customize them based on your device. Here we go…

Here you see the frequency response and phase— as you can see, it is fairly flat. And down here, we have our frequency response aligned to zero dB at 1 kHz, which is the standard way of displaying microphone response, and we can see our measured sensitivity is about -5.5dBv/Pa.

Now, let’s look at our alternative method—the Microphone Substitution Method.

In this method, rather than using an equalized speaker we first play our stepped sine sweep and  measure with a calibrated reference microphone. We then substitute our microphone under test and measure again. Next we subtract out the measurement of the reference microphone to get the raw response of our microphone under test. This removes the effect of the speaker’s response on the measurement.

We need to make sure we have the test microphone and the reference microphone in exactly the same position to minimize variance in the measurement.

If we look at our test sequence, we can see that we have two  acquisition steps, one for the reference mic and one for the mic under test, and a curve division step using the division operation in linear mode.

Let’s run the sequence.

It reminds me to make sure my reference mic is calibrated, which it is, and it prompts me to put it in front of the mouth simulator… let’s do that. We’re using a source speaker instead, but you can use a mouth simulator if you prefer, they both fulfil the same function.

It’s acquired the data, and now it’s asking me to put the microphone under test in its place… I’ll try to get it as close to the same spot as possible.

Let’s continue. And it’s acquired the data, performed the division, and here is our response.

This lower graph here shows the responses of the two microphones – the blue line is the reference mic and the orange line is our mic under test. And this top graph here shows the normalized frequency response including phase after the division.

You can see the sensitivity at 1 kHz is -6 dB V/ Pa, which aligns well with our first measurement using the equalized speaker measurement.

So there you have it – two simple methods for measuring microphones that ensure any non-linearities in your speaker don’t affect the measurement. And of course these are some of the most basic microphone measurements that you can do in SoundCheck. We can also measure directional response, acoustic overload point, intermodulation distortion, signal-to-noise ratio, embedded mics, MEMS mics and more. If you have any questions, or would like to see more microphone measurements, please get in touch with us – we’d be happy to demonstrate!

Bluetooth Earbud Test Measurement

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How do you measure Bluetooth earbuds? In this short video, we demonstrate how to measure Bluetooth earbuds using SoundCheck. You’ll see how to pair the earbuds, ensure a proper fit for accurate results, and configure SoundCheck for measurements including left and right earphone frequency response, THD, send response of the microphone, and distortion. This method can be used for any Bluetooth audio device, from true wireless earbuds to professional over-ear headphones.

Watch this Video Demo of Bluetooth Acoustic Measurements on Headphones / Earbuds

Learn More about How to Test Earbuds and Bluetooth Audio Measurements

Here are some additional resources:

Video Transcript: How to Test Bluetooth Earbuds

 In this short video I’m going to show you how simple it is to measure a pair of Bluetooth earbuds using SoundCheck and our free Bluetooth headphone sequence.

We’re going to use a head and torso simulator for the most realistic representation of how a human would hear. If this is outside your budget , you can use a simpler headphone test fixture or an ear coupler. I’ll use the Portland tool and die Bluetooth interface to pair to the earphones and AudioConnect 2 will connect and power the TEDs microphones inside the ear simulators, as well as the SCM mic that I used to calibrate the mouth simulator.

We’ll measure the left and right earphone frequency response, THD, send response of the microphone, and I’ll also show the difference between Left and Right earphones.

I already have SoundCheck up and running on my laptop, with our standard headphone measurement sequence that you can download from our website.

When I hit start, it asks if I want to pair my Bluetooth headphone… I’ll say ‘yes’ and turn on pairing mode … the white light is flashing …. and the Bluetooth Interface is searching for any devices that are open for pairing. We see the airpod pros here so I just hit OK…and once I’m connected the light turns green to show it’s paired.

Now the sequence is prompting to insert the earphones… I’ll put them in… and it wants  to check the seal. This is important to make sure there is a good fit – you’re going to need that to get good bass output. I’m going to say ‘yes’ and now the sequence is bringing up the signal generator playing pink noise in both ears. We can see the real time analyzer is showing the frequency response of both left and right ear. I have a good seal on my right but not on my left ear so I’ve got to adjust that a little to get a snug fit.

Now that I have a good seal on both ears, I can continue and we’re ready to measure.

We’re using a stepped sine in 12th octave frequency resolution from 20 Hertz to 20 kilohertz and we’ll see the frequency response as well as the diffuse field corrected response, which takes into account the ear simulators’ response on hats.

Here’s our results. We can see the frequency response – the green and red lines for left and right ear, and the diffuse field corrected, the purple line. Ideally this should be more or less flat when we subtract out the ear simulator, so this looks pretty good. 

Down below we see the difference between the left and right ear, the tracking. Ideally this would be zero, but we have a little notch here so there’s a small difference. On the bottom graph we have total harmonic distortion for the left and right ear.

Now it’s asking if I want to measure the microphone response otherwise known as the send response, and I say yes. We’ll hear pink noise played out of the mouth, and we will get the frequency response of the microphone 

So essentially I’m done now and it’s checking if I want to save the results. I also have information about the sensitivity of my earbuds and their microphone. That’s a very basic Bluetooth headphone test – there are many other measurements we can make on headphones and earbuds, and we’ll show you those in another video.

As you can see, measuring Bluetooth earbuds, or any type of Bluetooth headphone, is quick and easy with SoundCheck. If you’d like to give it a try, contact us for a demo!

HDMI Audio Measurements on a Soundbar

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How do you measure audio over HDMI? HDMI interfaces are common on soundbars and surround sound systems for the transmission of high definition audio and video. Some manufacturers sell expensive interfaces to connect the HDMI input of the device to the computer or test system, but with SoundCheck, there is a simpler and more cost-effective solution.

A simple $100 (approx) audio extractor, or switch, that you can buy online or from any electronics store is all that you need to add to your SoundCheck system to carry out a range of multi-channel audio tests on your device.

In this short video, we demonstrate how to perform HDMI audio measurements using SoundCheck. We show how to connect the soundbar via the switch, configure SoundCheck’s signal paths and calibration settings to support the HDMI connection, and make basic audio measurements including frequency response, THD, Rub & Buzz, and perceptual Rub & Buzz (ePRB). This method can be used for any HDMI-connected audio device, for example,  TVs, monitors, and home theater speakers.

Watch this Video Demo of HDMI Audio Measurements on a Soundbar

Learn More about How to Test Soundbars and HDMI Audio Measurements

Here are some additional resources on measuring soundbars:

Video Transcript: How to Test Soundbar Audio over HDMI

In this short video, I’ll show you how easy it is to test an HDMI-connected soundbar using SoundCheck. Typically the biggest challenge with testing soundbars is the HDMI connection, so that’s what we’re going to focus on here. The speaker tests themselves are quite simple – we’re just going to measure frequency response, THD and Rub & Buzz. If you’re interested in learning more about those measurements, check out our loudspeaker testing demo.

I’m going to use a basic test sequence that you can download from our website. This test sequence isn’t exclusive to soundbars  – it can be used to make audio measurements on any HDMI audio device, for example TVs, monitors, home theater speakers, etc. You can easily modify it for additional measurements and different devices.

Let’s start with the setup. As you can see from this schematic diagram, the equipment is simple: a computer running SoundCheck, a measurement microphone, and a simple low-cost 2-channel audio interface. 

The audio signal from the test computer is sent to the soundbar via a simple HDMI audio extractor, or switch, between the computer and the soundbar. This is an inexpensive component, maybe $100 or so, and it’s easy to find online or your local electronic store. This separates the audio from the HDMI signals so they can be output to an audio system such as an ARC or eARC soundbar, home theater or multimedia speaker. 

Because this is a basic test, we’re just going to use one microphone in the near field as you might for a production line quality control check. If you’re doing this in an R&D environment, you could measure at multiple locations with additional microphones and a multi-channel interface like the AmpConnect 621.

First, let’s configure our SoundCheck system’s Hardware and Calibration to support the HDMI connected device. We’ll connect the AudioConnect 2 using ASIO for input, and use the WASAPI driver option in SoundCheck to connect the HDMI soundbar. This provides multichannel support, more stable latency than WDM and lets multiple WASAPI devices be used simultaneously. 

Now we’ll configure the HDMI device. We’ll connect the computer’s HDMI into my HDMI extractor, and the HDMI output of that into the soundbar, then we’ll launch SoundCheck. Now, we’ll go to the hardware setup and configure the output channels. I’m going to add an HDMI channel here… let’s call it HDMI 1. I’ll select the WASAPI driver, then I’ll set the output for my HDMI device to digital. Lastly, I need to set the sample rate to match my input channel hardware…that’s 44.1 kilohertz for the AudioConnect 2. Now let’s repeat this for channel 2.

The WASAPI driver allows for multichannel input/output with SoundCheck. I also can easily adapt this test for a multichannel ATMOS system with an HDMI AVR receiver.

Next we need to create HDMI signal paths. Now we’re going to place the microphone, which I’ve already calibrated,  half a meter from the soundbar, centered between the speakers as we want it to be in the near field.  

Now let’s run the sequence.

Here’s our results. On this graph, we have the frequency response – that’s the blue line, and also the THD – that’s the orange line.

And on this second graph, we show Rub & Buzz, measured using both traditional and perceptual methods.

If we take a quick look at the test sequence you can see it’s really simple. We just play the test stimulus… you can see it’s a high to low sweep from 20 KHz to 100 Hertz. We analyze frequency response, THD, Rub & Buzz and Perceptual Rub and Buzz, and we display them on the screen.

So, as you can see, configuring SoundCheck for testing HDMI audio measurements is fast and easy. You just need to configure your HDMI channels first, then a test is implemented just the same way as any other audio test! This sequence I just used can be used as a base for audio measurements on any HDMI-connected device – just change the test parameters and limits as needed, and you can even add additional measurements.

I hope you found this demonstration  useful. If you’d like to learn more about HDMI audio measurements, or discuss your test application, please contact Listen or your local distributor. 

 

ABOUT LISTEN INC

Listen is a global leader in audio measurement and analysis, with over 30 years’ experience developing innovative test methods and algorithms. SoundCheck, our powerful software-based audio analyzer, pairs with our test interfaces, precision microphones, and accessories to offer a complete solution for testing loudspeakers, headphones, microphones, smart devices, communication systems, hearing aids, automotive audio, and more. Trusted by engineers worldwide, Listen delivers the accuracy and flexibility essential for modern audio testing, from the R&D lab to the production line.

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